DTV broadcasting in the United States of America has been done in accordance with broadcasting standards formulated by an industry consortium called the Advanced Television Systems Committee (ATSC). ATSC published a Digital Television Standard in 1995 that employed 8-level vestigial-sideband amplitude modulation of a single radio-frequency (RF) carrier wave. This DTV transmission system is referred to as 8-VSB. In the beginning years of the twenty-first century efforts were made to provide for more robust transmission of data over broadcast DTV channels without unduly disrupting the operation of so-called “legacy” DTV receivers already in the field. These efforts culminated in an ATSC standard directed to broadcasting digital television and digital data to mobile receivers being adopted on 15 Oct. 2009. This subsequent standard also used 8-level vestigial-sideband amplitude modulation of a single RF carrier wave, so the more robust transmission of data could be time-division multiplexed with the transmission of DTV signal to so-called “legacy” DTV receivers already in the field.
DTV broadcasting in Europe has employed coded orthogonal frequency-division multiplexing (COFDM) that employs a multiplicity of RF carrier waves closely spaced across each 6-, 7- or 8-MHz-wide television channel, rather than a single RF carrier wave per television channel. Adjacent carrier waves are orthogonal to each other. Successive multi-bit symbols are selected from a serial data stream and used to modulate respective ones of the multiplicity of RF carrier waves in turn, in accordance with a conventional modulation scheme—such as quaternary phase shift keying (QPSK) or quadrature amplitude modulation (QAM). QPSK is preferably DQPSK, using differential modulation that is inherently insensitive to slowly changing amplitude and phase distortion. DPSK simplifies carrier recovery in the receiver. Customarily, the QAM is either 16 or 64 QAM using square 2-dimensional modulation constellations. In actual practice, the RF carrier waves are not modulated individually. Rather, a single carrier wave is modulated at high symbol rate using QPSK or QAM. The resulting modulated carrier wave is then transformed in a fast inverse discrete Fourier transform (I-DFT) procedure to generate the multiplicity of RF carrier waves each modulated at low symbol rate.
In Europe, broadcasting to handset receivers is done using a system referred to as DVB-H. DVB-H (Digital Video Broadcasting—Handset) is a digital broadcast standard for the transmission of broadcast content to handset receivers, published in 2004 by the European Telecommunications Standards Institute (ETSI) and identified as EN 302304. DVB-H, as a transmission standard, specifies the physical layer as well as the elements of the lower protocol layers. It uses a power-saving technique based on the time-multiplexed transmission of different services. The technique, called “time-slicing”, allows substantial saving of battery power. Time-slicing facilitates soft hand-over as the receiver moves from network cell to network cell. The relatively long power-save periods may be used to search for channels in neighboring radio cells offering the selected service. Accordingly, at the border between two cells, a channel hand-over can be performed that is imperceptible by the user. Both the monitoring of the services in adjacent cells and the reception of the selected service data can utilize the same front-end tuner.
In contrast to other DVB transmission systems, which are based on the DVB Transport Stream adopted from the MPEG-2 standard, the DVB-H system is based on Internet Protocol (IP). The DVB-H baseband interface is an IP interface allowing the DVB-H system to be combined with other IP-based networks. Even so, the MPEG-2 transport stream is still used by the base layer. The IP data are embedded into the transport stream using Multi-Protocol Encapsulation (MPE), an adaptation protocol defined in the DVB Data Broadcast Specification. At the MPE level, DVB-H employs an additional stage of forward error correction called MPE-FEC, which is essentially (255, 191) transverse Reed-Solomon (TRS) coding. This TRS coding reduces the S/N requirements for reception by a handheld device by a 7 dB margin compared to DVB-T. The block interleaver used for the TRS coding creates a specific frame structure, called the “FEC frame”, for incorporating the incoming data of the DVB-H codec.
The physical radio transmission of DVB-H is performed according to the DVB-T standard and employs OFDM multi-carrier modulation. DVB-T employed coded orthogonal frequency division multiplexing (COFDM) in which an 8-MHz-wide radio-frequency (RF) channel comprises somewhat fewer than 2000 or somewhat fewer than 8000 evenly-spaced carriers for transmitting to stationary DTV receivers. DVB-T2, an upgrade of DVB-T proposed in 2011, further permits somewhat fewer than 4000 evenly-spaced carrier waves, better to accommodate transmitting to mobile receivers. These choices as to number of carrier waves are commonly referred to as 2K, 8K and 4K options.
Generally, DVB-H uses only a fraction (e.g., one quarter) of the digital payload capacity of the RF channel. Typically, consecutive time intervals referred to as “super-frames” are each composed of four consecutive frame intervals of like duration, three of which frame intervals are employed for DVB-T or DVB-T2 broadcasting to stationary DTV receivers. The fourth frame interval in each super-frame is divided into eight time-slice intervals that are employed for DVB-H broadcasting to mobile and handset DTV receivers.
COFDM has been considered for DTV broadcasting in the United States of America more than once, competing in the 1990's with 8-VSB and other single-carrier transmission systems for selection by ATSC as its Digital Television Standard. COFDM was considered as a replacement for 8-VSB at the time that the ATSC Digital Television Standard was updated to permit more robust transmissions for reception by mobile receivers. At that time any technical advantages of COFDM were over-ridden by the need not to obsolete DTV receivers already in the field, lest advertising-supported over-the-air DTV fail as a commercially viable business. However, reportedly COFDM is better adapted for use in single-frequency networks (SFNs) than is 8-VSB amplitude modulation, so COFDM is likely again to be considered for DTV broadcasting in the United States of America. The 2K, 8K and 4K options are retained in proposals for such DTV broadcasting, with bit rates being scaled back to suit the 6-MHz-wide RF channels used in the United States rather than the 8-MHz-wide RF channels used in Europe.
COFDM is able to overcome frequency-selective fading quite well, but reception will fail if there is protracted severe flat-spectrum fading. Such flat-spectrum fading is sometimes referred to as a drop-out in received signal strength. Such drop-out occurs when the receiving site changes such that a sole effective signal transmission path is blocked by an intervening hill or structure, for example. Because the signaling rate in the individual OFDM carriers is very low, COFDM receivers are capable of maintaining reception despite drop-outs that are only a fraction of a second in duration. However, drop-outs that last as long as a few seconds disrupt television reception perceptibly. Such protracted drop-outs are encountered in a vehicular receiver when the vehicle passes through a tunnel, for example. By way of further example of a protracted drop-out in reception, a stationary DTV receiver may briefly discontinue COFDM reception when receiver synchronization is momentarily lost during dynamic multipath reception conditions, as caused by aircraft flying over the reception site.
The ATSC standard directed to broadcasting digital television and digital data to mobile receivers used TRS coding that extended over eighty or a few more dispersed-in-time short time-slot intervals, rather than being confined to a single longer time-slot interval. A principal purpose of the TRS coding that extended over eighty or so time-slot intervals was overcoming occasional protracted drop-outs in received signal strength. Confining TRS coding to a single longer time-slot interval as done in DVB-H sacrifices such capability, but is advantageous in that error-correction is completed within a shorter time. This helps speed up changes in RF channel tuning, for example.
Iterative-diversity transmissions were proposed to facilitate alternative or additional techniques for dealing with flat-spectrum fading of 8-VSB signals. Some of these proposals were directed to separate procedures being used for decoding earlier and later transmissions of the same coded data to generate respective sets of data packets, each identified after such decoding either as being probably correct or probably incorrect. Corresponding data packets from the two sets were compared, and a further set of data packets was chosen from the ones of the compared data packets more likely to be correct. A. L. R. Limberg proposed delaying earlier transmissions of concatenated convolutionally coded (CCC) data so as to be contemporaneously available with later transmissions of similar CCC data, then decoding the contemporaneous CCC data with respective turbo decoders that exchanged information concerning soft data bits to secure coding gain. These techniques, although comparatively robust in regard to overcoming additive White Gaussian noise (AWGN), reduce code rate by a factor of three, as compared to non-repeated transmissions with simple one-half-rate convolutional coding. Also, these techniques require more delay memory for the earlier transmitted data than does implementation of iterative-diversity reception at the transfer-stream (TS) data-packet level, owing to the parity bits of the FEC coding of the data also having to be delayed.
The parallel iterative operation of two turbo decoders consumes more power than is desirable, particularly in battery-powered receivers. Maximal-ratio code combining is a technique that has been used for combining similar transmissions from a plurality of transmitters in multiple-input/multiple-output (MIMO) networks. Searching for a way to avoid parallel iterative operation of two turbo decoders, A. L. R. Limberg considered the use of maximal ratio code combining of later transmissions of CCC with earlier similar CCC transmissions from the same 8-VSB transmitter. The hope was that a combined signal would be generated that could be decoded by iterative operation of a single turbo decoder. One problem encountered when trying to implement such an approach in 8-VSB broadcasting is that the coding of M/H-service data is not independent of the coding of main-service data. The inner convolutional coding of the M/H signal is part of a one-half-rate convolutional coding that intersperses main-service signal components with M/H-service signal components. Accordingly, practically considered, the inner convolutional coding of the later transmissions of CCC and the inner convolutional coding of the delayed earlier transmissions of CCC still have to be decoded separately. The outer convolutional coding of the M/H signal is affected by the pre-coding of the most-significant bits of 8-VSB symbols responding to main-service data interspersed among the most-significant bits of 8-VSB symbols responding to M/H-service data. There are also some problems with measuring the energies of the later transmissions of CCC and the delayed earlier transmissions of CCC to provide the information needed for weighting these transmissions for maximal-ratio code combining.
In a replacement system for DTV broadcasting in the United States of America that uses COFDM of a plurality of carrier waves, the FEC coding of main-service data and the FEC coding of M/H-service data can be kept independent of each other. Also, the inclusion of unmodulated carrier waves among the COFDM carrier waves facilitates measurements of their total root-mean-square (RMS) energy in later transmissions and in earlier transmissions of similar data to provide the information needed to weight later and delayed earlier transmissions appropriately for maximal-ratio code combining. The COFDM is based on complex-number coordinates of two-dimensional symbol constellations of quadrature amplitude modulation (QAM)
Iterative-diversity reception implemented at the transfer-stream (TS) data-packet level does not require as much delay memory for the earlier transmitted data as delaying complete earlier transmissions to be contemporaneous with later transmissions of the same data. This is because the redundant parity bits associated with FEC coding contained in those complete earlier transmissions is removed during its decoding and so do not need to be delayed. However, implementation of iterative-diversity reception at the TS data-packet level sacrifices the substantial coding gain that can be achieved by decoding delayed earlier transmissions concurrently with later transmissions of similar data. Implementation of iterative-diversity reception at the TS data-packet level is also incompatible with code-combining of delayed earlier transmissions and later transmissions of similar data being used to improve signal-to-noise ratio (SNR).
Time-sliced reception can substantially reduce the amount of delay memory required for iterative-diversity reception of DTV signals transmitted using COFDM. Such reduction in the size of delay memory also conserves the operating power that would otherwise be consumed by the eliminated memory. If each service broadcast to stationary DTV receivers occupies no more than one frame out of four in a super-frame, time-sliced reception can reduce by a factor of four the amount of delay memory that a stationary DTV receiver requires for iterative-diversity reception. If each service broadcast to M/H receivers occupies no more than 1/32 of a super-frame, time-sliced reception can reduce by a factor of thirty-two the amount of delay memory that an M/H receiver requires for iterative-diversity reception.
DVB-T2 employs low-density parity check (LDPC) coding as forward-error-correction (FEC) coding to help overcome intersymbol interference (ISI) and other AWGN, rather than using convolutional coding, product coding, or concatenated convolutional coding (CCC). An LDPC code is based on an H matrix containing a low count of ones. Encoding uses equations derived from the H matrix to generate the parity check bits. Decoding is accomplished using these equations with “soft-decisions” as to transmitted symbols to generate new estimates of the transmitted symbols. This process is repeated in an iterative manner resulting in a powerful decoder. Like parallel concatenated convolutional coding (PCCC), LDPC codes are subject to error floors. Outer coding, such as Bose-Chaudhuri-Hocquenghem (BCH) coding, can be added to LDPC technology to lower the error floor. The BCH coding can be Reed-Solomon (RS) coding, for example. Reportedly, LDPC coding provides AWGN performance that can approach the Shannon Limit even more closely than PCCC.
In US-2010-0293433-A1 published 18 Nov. 2010 with the title “Burst-error correction methods and apparatuses for wireless digital communications systems” A. L. R. Limberg described the data in the initial-transmission component of an iterative-diversity 8-VSB AM transmission differing from the data bits in the final-transmission component, the data bits in each of the components of that iterative-diversity transmission being the ONEs' complements of the data bits in the other one of the components. ONEs' complementing the data bits in the initial-transmission component of an iterative-diversity transmission tends to increase the number of ONEs therein when the original set of data bits is sparsely populated by ONEs. Accordingly, parallel concatenated convolutional coding (PCCC) generated from that data is less likely to be sparsely populated by ONEs. Sparse population of ONEs in PCCC coding is known to result in poorer reception via Rayleigh channels.
In the prior art one way that designers attempted to avoid the problem of sparse population of ONEs in convolutional coding or in PCCC was by using recursive systematic convolutional (RSC) coding, rather than non-systematic convolutional (NSC) coding that does not involve recursion. The recursion in RSC coding tends to generate ONEs in its parity bits, despite a limited number of ONEs in its data bits. This way of trying to avoid the problem of sparse population of ONEs depends upon the component codes in concatenated coding using recursion, which is not true of certain codes such as LDPC codes. ONEs' complementing the data bits when generating one of the component codes of concatenated coding will provide relief from sparsity of ONEs even when using coding methods that do not employ recursion.
In the prior art another way that designers attempted to avoid the problem of sparse population of ONEs in PCCC was by interleaving data bits for one of the component convolutional coding procedures to disperse ONEs differently than the data bits involved in the other of the component convolutional coding procedures. This had the further beneficial effect that additive white Gaussian noise (AWGN) would affect the component convolutional codes differently, even though the data bits were transmitted only once. The effects of AWGN on the parity bits in the component convolutional codes is different, the bit-interleaving providing a degree of temporal diversity between the component convolutional codes that aids in iterative decoding procedures called “turbo decoding”.
The bit-interleaving of data in one component of convolutional coding in prior-art PCCC introduces undesirable delay into the soft-input/soft-output decoding of that component convolutional coding. This delay is introduced during the de-interleaving of the bit-interleaved results of the SISO decoding of that component convolutional coding. This delay is introduced in each cycle of turbo decoding in the receiver, and appreciably slows the iterative decoding procedures in aggregate.